[asterisk-commits] oej: trunk r48277 - in /trunk: include/asterisk/
main/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 5 13:39:13 MST 2006
Author: oej
Date: Tue Dec 5 14:39:13 2006
New Revision: 48277
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48277
Log:
Doxygen updates
Modified:
trunk/include/asterisk/channel.h
trunk/include/asterisk/rtp.h
trunk/main/rtp.c
Modified: trunk/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/channel.h?view=diff&rev=48277&r1=48276&r2=48277
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Tue Dec 5 14:39:13 2006
@@ -81,7 +81,38 @@
\arg translate.h - Transcoding support functions
\arg \ref channel_drivers - Implemented channel drivers
\arg \ref Def_Frame Asterisk Multimedia Frames
-
+ \arg \ref Def_Bridge
+
+*/
+/*! \page Def_Bridge Asterisk Channel Bridges
+
+ In Asterisk, there's several media bridges.
+
+ The Core bridge handles two channels (a "phone call") and bridge
+ them together.
+
+ The conference bridge (meetme) handles several channels simultaneosly
+ with the support of an external timer (zaptel timer). This is used
+ not only by the Conference application (meetme) but also by the
+ page application and the SLA system introduced in 1.4.
+ The conference bridge does not handle video.
+
+ When two channels of the same type connect, the channel driver
+ or the media subsystem used by the channel driver (i.e. RTP)
+ can create a native bridge without sending media through the
+ core.
+
+ Native briding can be disabled by a number of reasons,
+ like DTMF being needed by the core or codecs being incompatible
+ so a transcoding module is needed.
+
+References:
+ - ast_channel_early_bridge()
+ - ast_channel_bridge()
+ - app_meetme.c
+ - \ref AstRTPbridge
+ - ast_rtp_bridge()
+ - \ref Def_Channel
*/
/*! \page AstFileDesc File descriptors
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?view=diff&rev=48277&r1=48276&r2=48277
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Tue Dec 5 14:39:13 2006
@@ -48,7 +48,10 @@
/*! Maximum RTP-specific code */
#define AST_RTP_MAX AST_RTP_CISCO_DTMF
+/*! Maxmum number of payload defintions for a RTP session */
#define MAX_RTP_PT 256
+
+#define FLAG_3389_WARNING (1 << 0)
enum ast_rtp_options {
AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
@@ -65,6 +68,8 @@
struct ast_rtp;
+/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem
+*/
struct ast_rtp_protocol {
/*! Get RTP struct, or NULL if unwilling to transfer */
enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
@@ -78,8 +83,7 @@
};
-#define FLAG_3389_WARNING (1 << 0)
-
+/*! RTP callback structure */
typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
/*!
@@ -122,11 +126,13 @@
struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
+/*! Destroy RTP session */
void ast_rtp_destroy(struct ast_rtp *rtp);
void ast_rtp_reset(struct ast_rtp *rtp);
-void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
+/*! Stop RTP session, do not destroy structure */
+void ast_rtp_stop(struct ast_rtp *rtp);
void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
@@ -189,10 +195,18 @@
/*! \brief Enable STUN capability */
void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
+/*! \brief Send STUN request (??) */
+void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
+
+/*! \brief The RTP bridge.
+ \arg \ref AstRTPbridge
+*/
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
+/*! \brief Register an RTP channel client */
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
+/*! \brief Unregister an RTP channel client */
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
@@ -201,22 +215,22 @@
having to send a re-invite later */
int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
-void ast_rtp_stop(struct ast_rtp *rtp);
-
-/*! \brief Return RTCP quality string */
-char *ast_rtp_get_quality(struct ast_rtp *rtp);
+
/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
int ast_rtcp_send_h261fur(void *data);
-void ast_rtp_init(void);
-
-int ast_rtp_reload(void);
-
+char *ast_rtp_get_quality(struct ast_rtp *rtp); /*! \brief Return RTCP quality string */
+void ast_rtp_init(void); /*! Initialize RTP subsystem */
+int ast_rtp_reload(void); /*! reload rtp configuration */
+
+/*! Set codec preference */
int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
+/*! Get codec preference */
struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
+/*! get format from predefined dynamic payload format */
int ast_rtp_codec_getformat(int pt);
/*! \brief Set rtp timeout */
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=48277&r1=48276&r2=48277
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Tue Dec 5 14:39:13 2006
@@ -3142,7 +3142,34 @@
/*! \brief Bridge calls. If possible and allowed, initiate
re-invite so the peers exchange media directly outside
- of Asterisk. */
+ of Asterisk.
+*/
+/*! \page AstRTPbridge The Asterisk RTP bridge
+ The RTP bridge is called from the channel drivers that are using the RTP
+ subsystem in Asterisk - like SIP, H.323 and Jingle/Google Talk.
+
+ This bridge aims to offload the Asterisk server by setting up
+ the media stream directly between the endpoints, keeping the
+ signalling in Asterisk.
+
+ It checks with the channel driver, using a callback function, if
+ there are possibilities for a remote bridge.
+
+ If this fails, the bridge hands off to the core bridge. Reasons
+ can be NAT support needed, DTMF features in audio needed by
+ the PBX for transfers or spying/monitoring on channels.
+
+ If transcoding is needed - we can't do a remote bridge.
+ If only NAT support is needed, we're using Asterisk in
+ RTP proxy mode with the p2p RTP bridge, basically
+ forwarding incoming audio packets to the outbound
+ stream on a network level.
+
+ References:
+ - ast_rtp_bridge()
+ - ast_channel_early_bridge()
+ - ast_channel_bridge()
+*/
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
{
struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
More information about the asterisk-commits
mailing list